Personally, I do not need digital outputs from the G2. I do not have "golden ears", so it would not surprise me at all if I could not hear a difference between the G2's DACs and the current "state of the art" external DACs mixed out to the monitors. I would guess that the only advantage for a G2 digital output after this would be for post digital audio processing (to eliminate a D to A and A to D conversion and to keep the G2 signal in the digital domain until all digital audio processing was completed).

Just my opinion (but, I could be wrong) - I think that Clavia's efforts to make the G2 "sound better" might be directed more towards:

1. making the G2 filters sound more like analog filters (it is obvious even to my bad ears that the G2 filters have a "digital grit", especially at high resonance and cutoff settings).

2. improving the anti-aliasing algorithms.

As is usually the case though (as with all digital processing), the Clavia software engineers may have already reached a point of compromise with the current G2 hardware - that to keep a reasonable amount of DSP power available for voice polyphony count, they had to give up on using the power for better sounding filters and/or anti-aliasing algorithms._________________varice

Velocity should of course take the Fletcher-Munson equal loudness curves into account. After all, the final chain in the audio chain is the human brain, which is far from linear.

With you in principle but not in implementation here. I think velocity curves should take human ergonomics, key weight and player strength into acount. Where paremeters like velocity are linked to things affecting frequency content then by all means, yes, link the resulting velocity value through that curve for that parameter, but not blindly for every parameter in sight._________________Kassen

With you in principle but not in implementation here. I think velocity curves should take human ergonomics, key weight and player strength into acount. Where paremeters like velocity are linked to things affecting frequency content then by all means, yes, link the resulting velocity value through that curve for that parameter, but not blindly for every parameter in sight.

Sorry, I fail to see this "blindly for every parameter in sight" in the implementation suggested in my post. So, I suppose you hint on a more general guideline here, right?

Let me elaborate a little on this "Fletcher-Munson / Robinson-Dadson / ISO Recommendation 226" thingy, as many might profit. It shows that when a sound is played at a higher sound pressure level not all frequencies are perceived to be 'louder' in the same amounts as would the increase in SPL suggest. E.g. the middle frequency range of the 'ear' is much more sensitive than the low or very high frequencies for changes in SPL. When velocity on a synth is used exclusively to control only amplitude there is actually an incorrect perception of increase in loudness. (Note that I talk about human perception, or how the brain interpretes sonic stimulae) This is also demonstrated very well by samplers using the same sample for a single note at different 'velocity' values, this sounds totally unnatural. The perception gets more natural if the middle frequencies are affected more in amplitude, compared to the higher or lower frequencies, when the velocity value is higher. The reason why this is important is that sounds in a mix have a relative loudness compared to the other sounds in a mix. Note that it is unknown at what real sound pressure level the recorded mix will be played later in someone's living room. So, the human mind needs to be 'tricked' into getting the correct loudness feel for different actual sound pressure levels. And luckily for the modern producer/sound engineer the human mind gets tricked easily and the tools to do so are there.

So, if two instruments are set against each other in a mix, and the idea of varying loudness of one instrument against the other is used in the mix, it is not at all sufficient to just control the amplitude of the sound that varies in loudness. Lets say that one instrument is increased in loudness in a way that it should sound twice as loud compared to the other instrument. This would mean that at 1 kHz the extra amplification caused by the velocity should be about 10 dB while at 50 Hz it would be only some 6 dB. In practice the real sonic parameter controlled by velocity is 'presence in the mix' or simply 'presence'. Having velocity control presence works much better as just having velocity control the overall amplitude of a sound. The idea of presence is directly derived from the ISO 226 graph or better how the two spectra of the two sounds in the mix relate to each other when their spectra are set out on the ISO 226 curves.
In practice it means that the relative amplitude of an instrument in a mix doesn't have to vary much, but the effect of playing louder and softer than the other instruments is for a big part created by how the energy level in the 1khz to 4khz area changes in relation to the energy level of the area between 200 Hz and 1kHz. The simplest way to imitate this behaviour is to use a peak eq with a wide band centered around 2 kHz and balance the velocity value between the straight overall amplitude and the peak level value of the eq. The ISO 226 graphs might at first sight seem to suggest a peak at 3kHz as there is a strong dip there, but the wide bell curve of the eq approximates the ISO 226 curves better at 2kHz (the ideal curve is much more complex to create).
So, if in the mix one sound gets an increase of 4 dB at 1 kHz compared to the level at 50 Hz (which is the 10 dB for 1 kHz minus the 6 dB for the 50Hz of the earlier example) it subjectively feels like the sound is played louder, as the subconcious mind thinks "hey, something changed according to ISO 226 so the guy must be hammering more intensely". And as it is general knowledge that people preferrably only hear what they like to hear all is hunky dory.
It also means that it hardly pays to control the amplitude of a bass sound by the velocity parameter, instead it pays to control the relative loudness of the harmonics in the bass signal compared to the loudness level of the bass fundamental. Which I guess is something that about everyone on this forum knows very well from mixing practice. For sounds with higher pitches the same underlying principle is at work, but here 'presence' means to find the right balance between overall amplitude and spectrum.

And Kassen, of course, for delicate and complex physical models, controlled by delicate and complex play controllers, there might be more play data available than the rudimentary key velocity measurement value of the plastic synth keyboard. Meaning that approaches could take different angles more in accordance with what is actually modelled.
But the world is not yet very rich in elaborate play controllers and physical models to play them, so just apply all I just wrote to all that "By Jove, this has all done before and blaah" simplistic stuff. Just to keep the fun in what us guys and gals do and listen to. As I don't think you would want your lunch to leave the same route it entered, right? Ok, I shut up...

Looking over the specs of the CS4392 DAC it appears to have 192 kHz capability, DSD capability and selectable digital filter roll-offs. Once a DAC has been implemented in the hardware, can any of these features be enabled or disabled through software?

Velocity should of course take the Fletcher-Munson equal loudness curves into account. After all, the final chain in the audio chain is the human brain, which is far from linear.

Yup, this is one very interesting suggestion. I would love this in DAWs too, implemented in some sort of channel level smart EQ thingie. Perhaps it makes more sense in a DAW though. On the other hand, you can pretty much use the G2 for a complete piece of music and in that context, having a Fletcher-Munson or Robinson/Dadson ( pretty much means the same thing these days) mode might be just what the doctor ordered.

Perceptive loudness does of course also depend a lot on how audio events are understood. This means that audio stream formation theory is relevant too, but then that would be in this context a direct discussion of the music vs. composition.. so taking that one further would have to be done inside the Composition forum I guess.

If I remember my textbooks correctly the Fletcher Munson work was done in the 1930s. Parts of the work resembles a bit how the IEC colour space work was done. Pretty cool.

Yup, we all know the basics of how this works, but we still haven´t any smart tools available. We are mostly still using EQ and manually adjusting channels across the board.

Quote:

Perceptual Audio Demonstrations
A number of demonstrations were created to help instructors teach their students about the human auditory system and its limitations. In order to understand the demonstrations and the concepts behind them, some background material and detailed explanations are given for each topic.

Cebec: DSD is impossible to implement. It is a streaming system that's an alternative to PCM data, where the sample rate actually changes to recreate the audio, rather than the amplitude of a sample. It's used (I believe) in the newer SACD format and other optical formats, because it can be directly encoded to disc.

Also, there's no possibility of exporting 192kHz data from the 96kHz that the G2 provides.

Last, Jan is right: if the processor isn't connected to the DAC's "control port", nothing can change. However, I think it is connected, both by looking at the pc board, and by the fact that the dac isn't running in "Slow" mode (<50kHz) (which I it seems is the only mode the dac can run in stand-alone).
.

However, I think it is connected, both by looking at the pc board, and by the fact that the dac isn't running in "Slow" mode (<50kHz) (which I it seems is the only mode the dac can run in stand-alone).
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Did I misinterpret the description of the "m" inputs ? Thought those should do it ...

You're right Jan. I read the part that said "In stand-alone mode, only the fast roll-off filter is available". That stuck in my head as the only other thing I'd want to change if I could change anything.

Also, looking at the spec & the PCB, there is an analog filter placed *after* the DAC. It would be simple (and relatively low-risk) to piggy-back components on the PCB to modify what the response of that filter is.

Any takers on what it "should" be for a warmer non-linear freq. response?

I went back to your original post jk, earlier in this thread:
Forgive me for not being up to speed on some of these issues, but I'm trying to understand.
Is that hump we're seeing on the FFTs of the G2s DACs the "Double Speed (Fast) Passband Ripple" illustrated on page 29 of the 4392 datasheet? That bump's not visible on the graph of the "Double Speed (Slow) Passband Ripple" on page 30.
So, if the mode could be changed to Double Speed (Slow) we'd eliminate that HF noise hump, correct? Or are we looking at preemphasis or neither?
Would native digital deemphasis help? Like with an EQ plug-in using a High Shelf?

The hump is above 1/2 the sampled frequency, so it's either aliased from somewhere else, or has nothing to do with the DAC, and is just plain noise (most likely). Besides, it is very inaudible (-80db at inauduble frequencies). So, I don't think there's anything we could do with the DAC to effect that stuff. De-emphasis (I don't think) is on, since it can't be enabled at 96kHz.

Also, I tried Rob's suggestions for shaping a synth's output by using a (digital) 31 band EQ. I used much smaller values for the attenuation and tweaked it quite a bit since the sound was pretty dark when using -1 to -3 dB. Perhaps this is due to personal taste and the differences between EQs (analog or digital).

I'm looking forward to you talk(s) at the conference, Rob. Can you or anyone else recommend any sources online that discuss, in more detail, the applications of psychacoustics to digital audio recording like you've touched upon? I'd like to try applying psychoacoustics via EQ to the G2s signal...

It would be great to have a digital out in the nord modular. (SPDIF and/or COAX)

Yeah, I wish it had ADAT or an option bord for ADAT I/O. It is good to still have things to dream about, maybe a future G3 could have one of these options.

Here in germany sits a company that did all kind of digi out retrofits... 8 digital outs for akai samplers for example...with wordclock input.... aso...
They sit in Hamburg and i am sure that what was able to do with a S 1100 is possible with a G2 aswell... But it will be expensiv to ask for such a custom modification.. You have to sync the clock and need to connect and convert to the digital out busses of the G2...theoretical possible

I could do something like this, but it would require you to ship your G2 back & forth to Denver, solder onto the G2's main board, and drill hole(s) in the back for the S/PDIF or ADAT connector. Which voids any warranty you'd have on the G2.

I'd have to make a board to handle the new circuitry (not much!). Maybe I'll do this anyway, and sell it as a kit? Then you could have the mods done locally or by yourself. I don't think I'd do it unless people where lined up.

i probably can't afford anything like this in the near future, and probably not while my G2s under warranty, but I will consider it if it's 24/96 out. and, if not on my keyboard, maybe a future Engine... anyway, count me interested.

An old thread, I know, but I'm rewiring my studio again, and providing I/O to the G2 is always a bitch, especially when I would like to bypass all the A/D/A conversion done in my studio (still).

On another hand, here's some possibilities for balanced outputs, which would be nearly as good as a full digital option (certainly better than single-ended outs it has!). I had come across these articles recently at the Jensen website (nice transformers), and the reference to the Crystal Semiconductor DACs made me think of the G2. Maybe these could do good on the G2--

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